[asterisk-commits] twilson: trunk r98988 - in /trunk: configs/ doc/tex/ res/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 16 21:09:33 CST 2008
Author: twilson
Date: Wed Jan 16 21:09:32 2008
New Revision: 98988
URL: http://svn.digium.com/view/asterisk?view=rev&rev=98988
Log:
Update res_phoneprov to default to setting the SERVER variable to the IP
the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf. I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.
Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport. Tested on Linux and OS X.
Modified:
trunk/configs/phoneprov.conf.sample
trunk/doc/tex/phoneprov.tex
trunk/res/res_phoneprov.c
Modified: trunk/configs/phoneprov.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/phoneprov.conf.sample?view=diff&rev=98988&r1=98987&r2=98988
==============================================================================
--- trunk/configs/phoneprov.conf.sample (original)
+++ trunk/configs/phoneprov.conf.sample Wed Jan 16 21:09:32 2008
@@ -1,9 +1,14 @@
[general]
-;serveraddr=192.168.1.1 ; Address to send to the phone to use as server address.
-serveriface=eth0 ; Same as above, except an ethernet interface.
- ; Useful for when the interface uses DHCP.
- ; There is no default for either of the above, and only one should be set.
-serverport=5060 ; Port to send to the phone to use as server port. Default is 5060.
+; The default behavior of res_phoneprov will be to set the SERVER template variable to
+; the IP address that the phone uses to contact the provisioning server and the
+; SERVER_PORT variable to the bindport setting in sip.conf. Unless you have a very
+; unusual setup, you should not need to set serveraddr, serveriface, or serverport.
+
+;serveraddr=192.168.1.1 ; Override address to send to the phone to use as server address.
+;serveriface=eth0 ; Same as above, except an ethernet interface.
+ ; Useful for when the interface uses DHCP and the asterisk http
+ ; server listens on a different IP than chan_sip.
+;serverport=5060 ; Override port to send to the phone to use as server port.
default_profile=polycom ; The default profile to use if none specified in users.conf
; You can define profiles for different phones specifying what files to register
Modified: trunk/doc/tex/phoneprov.tex
URL: http://svn.digium.com/view/asterisk/trunk/doc/tex/phoneprov.tex?view=diff&rev=98988&r1=98987&r2=98988
==============================================================================
--- trunk/doc/tex/phoneprov.tex (original)
+++ trunk/doc/tex/phoneprov.tex Wed Jan 16 21:09:32 2008
@@ -21,20 +21,26 @@
\begin{verbatim}
[general]
;serveriface=eth0
-serveraddr=192.168.1.1
-serverport=5060
+;serveraddr=192.168.1.1
+;serverport=5060
default_profile=polycom
\end{verbatim}
\end{astlisting}
-There are two choices for setting the SERVER variable. If the IP address of the server is
-known, or the hostname resolvable by the phones, the appropriate \textbf{serveraddr}
-value should be set. Alternatively, the network interface that the server listens on can
-be set by specifying a \textbf{serveriface} and SERVER will be set to the IP address of
-that interface. Only one of these options should be set.
-
-The SERVER\_PORT variable is set by setting the \textbf{serverport}. If serverport is
-not specified, it is set to a default value of 5060.
+By default, res\_phoneprov will set the SERVER variable to the IP address on the server
+that the requesting phone uses to contact the asterisk HTTP server. The SERVER\_PORT
+variable will default to the \textbf{bindport} setting in sip.conf.
+
+Should the defaults be insufficient, there are two choices for overriding the default
+setting of the SERVER variable. If the IP address of the server is known, or the hostname
+resolvable by the phones, the appropriate \textbf{serveraddr} value should be set.
+Alternatively, the network interface that the server listens on can be set by specifying a
+\textbf{serveriface} and SERVER will be set to the IP address of that interface. Only one
+of these options should be set.
+
+The default SERVER\_PORT variable can be overridden by setting the \textbf{serverport}.
+If \textbf{bindport} is not set in \path{sip.conf} and serverport is not specified, it
+is set to a default value of 5060.
Any user set for auto-provisioning in users.conf without a specified profile will be
assumed to belong to the profile set with \textbf{default\_profile}.
Modified: trunk/res/res_phoneprov.c
URL: http://svn.digium.com/view/asterisk/trunk/res/res_phoneprov.c?view=diff&rev=98988&r1=98987&r2=98988
==============================================================================
--- trunk/res/res_phoneprov.c (original)
+++ trunk/res/res_phoneprov.c Wed Jan 16 21:09:32 2008
@@ -153,7 +153,7 @@
};
char global_server[80] = ""; /*!< Server to substitute into templates */
-char global_serverport[6] = "5060"; /*!< Server port to substitute into templates */
+char global_serverport[6] = ""; /*!< Server port to substitute into templates */
char global_default_profile[80] = ""; /*!< Default profile to use if one isn't specified */
/*! \brief List of global variables currently available: VOICEMAIL_EXTEN, EXTENSION_LENGTH */
@@ -395,6 +395,23 @@
if (file)
ast_free(file);
goto out500;
+ }
+
+ /* Unless we are overridden by serveriface or serveraddr, we set the SERVER variable to
+ * the IP address we are listening on that the phone contacted for this config file */
+ if (ast_strlen_zero(global_server)) {
+ struct sockaddr name;
+ socklen_t namelen = sizeof(name);
+ int res;
+
+ if ((res = getsockname(ser->fd, &name, &namelen)))
+ ast_log(LOG_WARNING, "Could not get server IP, breakage likely.\n");
+ else {
+ struct ast_var_t *var;
+
+ if ((var = ast_var_assign("SERVER", ast_inet_ntoa(((struct sockaddr_in *)&name)->sin_addr))))
+ AST_LIST_INSERT_TAIL(route->user->headp, var, entries);
+ }
}
pbx_substitute_variables_varshead(route->user->headp, file, tmp, bufsize);
@@ -698,10 +715,11 @@
if (!ast_strlen_zero(global_server)) {
if ((var = ast_var_assign("SERVER", global_server)))
AST_LIST_INSERT_TAIL(user->headp, var, entries);
- if (!ast_strlen_zero(global_serverport)) {
- if ((var = ast_var_assign("SERVER_PORT", global_serverport)))
- AST_LIST_INSERT_TAIL(user->headp, var, entries);
- }
+ }
+
+ if (!ast_strlen_zero(global_serverport)) {
+ if ((var = ast_var_assign("SERVER_PORT", global_serverport)))
+ AST_LIST_INSERT_TAIL(user->headp, var, entries);
}
/* Append profile variables here, and substitute variables on profile
@@ -736,20 +754,27 @@
/* \brief Parse config files and create appropriate structures */
static int set_config(void)
{
- struct ast_config *phoneprov_cfg, *users_cfg;
+ struct ast_config *cfg;
char *cat;
struct ast_variable *v;
struct ast_flags config_flags = { 0 };
- if (!(phoneprov_cfg = ast_config_load("phoneprov.conf", config_flags))) {
+ /* Try to grab the port from sip.conf. If we don't get it here, we'll set it
+ * to whatever is set in phoneprov.conf or default to 5060 */
+ if ((cfg = ast_config_load("sip.conf", config_flags))) {
+ ast_copy_string(global_serverport, S_OR(ast_variable_retrieve(cfg, "general", "bindport"), "5060"), sizeof(global_serverport));
+ ast_config_destroy(cfg);
+ }
+
+ if (!(cfg = ast_config_load("phoneprov.conf", config_flags))) {
ast_log(LOG_ERROR, "Unable to load config phoneprov.conf\n");
return -1;
}
cat = NULL;
- while ((cat = ast_category_browse(phoneprov_cfg, cat))) {
+ while ((cat = ast_category_browse(cfg, cat))) {
if (!strcasecmp(cat, "general")) {
- for (v = ast_variable_browse(phoneprov_cfg, cat); v; v = v->next) {
+ for (v = ast_variable_browse(cfg, cat); v; v = v->next) {
if (!strcasecmp(v->name, "serveraddr"))
ast_copy_string(global_server, v->value, sizeof(global_server));
else if (!strcasecmp(v->name, "serveriface")) {
@@ -761,28 +786,26 @@
else if (!strcasecmp(v->name, "default_profile"))
ast_copy_string(global_default_profile, v->value, sizeof(global_default_profile));
}
- if (ast_strlen_zero(global_server))
- ast_log(LOG_WARNING, "No serveraddr/serveriface set in phoneprov.conf. Breakage likely.\n");
} else
- build_profile(cat, ast_variable_browse(phoneprov_cfg, cat));
- }
-
- ast_config_destroy(phoneprov_cfg);
-
- if (!(users_cfg = ast_config_load("users.conf", config_flags))) {
+ build_profile(cat, ast_variable_browse(cfg, cat));
+ }
+
+ ast_config_destroy(cfg);
+
+ if (!(cfg = ast_config_load("users.conf", config_flags))) {
ast_log(LOG_WARNING, "Unable to load users.cfg\n");
return 0;
}
cat = NULL;
- while ((cat = ast_category_browse(users_cfg, cat))) {
+ while ((cat = ast_category_browse(cfg, cat))) {
const char *tmp, *mac;
struct user *user;
struct phone_profile *profile;
struct ast_var_t *var;
if (!strcasecmp(cat, "general")) {
- for (v = ast_variable_browse(users_cfg, cat); v; v = v->next) {
+ for (v = ast_variable_browse(cfg, cat); v; v = v->next) {
if (!strcasecmp(v->name, "vmexten")) {
if ((var = ast_var_assign("VOICEMAIL_EXTEN", v->value)))
AST_LIST_INSERT_TAIL(&global_variables, var, entries);
@@ -797,15 +820,15 @@
if (!strcasecmp(cat, "authentication"))
continue;
- if (!((tmp = ast_variable_retrieve(users_cfg, cat, "autoprov")) && ast_true(tmp)))
+ if (!((tmp = ast_variable_retrieve(cfg, cat, "autoprov")) && ast_true(tmp)))
continue;
- if (!(mac = ast_variable_retrieve(users_cfg, cat, "macaddress"))) {
+ if (!(mac = ast_variable_retrieve(cfg, cat, "macaddress"))) {
ast_log(LOG_WARNING, "autoprov set for %s, but no mac address - skipping.\n", cat);
continue;
}
- tmp = S_OR(ast_variable_retrieve(users_cfg, cat, "profile"), global_default_profile);
+ tmp = S_OR(ast_variable_retrieve(cfg, cat, "profile"), global_default_profile);
if (ast_strlen_zero(tmp)) {
ast_log(LOG_WARNING, "No profile for user [%s] with mac '%s' - skipping\n", cat, mac);
continue;
@@ -816,7 +839,7 @@
continue;
}
- if (!(user = build_user(users_cfg, cat, mac, profile))) {
+ if (!(user = build_user(cfg, cat, mac, profile))) {
ast_log(LOG_WARNING, "Could not create user %s - skipping.\n", cat);
continue;
}
@@ -832,7 +855,7 @@
AST_RWLIST_UNLOCK(&users);
}
- ast_config_destroy(users_cfg);
+ ast_config_destroy(cfg);
return 0;
}
More information about the asterisk-commits
mailing list